Commit Graph

4868 Commits

Author SHA1 Message Date
ef0d76ae83 Add more VP9 header correctness check in RtpFrameReferenceFinder
Bug: chromium:1049129
Change-Id: I133673d86aadd6a87b3420a04bbf45ed53841a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168240
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30466}
2020-02-06 08:39:44 +00:00
78c7c5247c Revert "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""""
This reverts commit 703a5d76d9ba8e7984509cc7bf70fb4ed84ef6be.

Reason for revert: Breaks a downstream project. I will notify when it is possible to reland.

Original change's description:
> Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""
> 
> This is a reland of af51be7869994a299451e22e6382ae641767b26d
> 
> Original change's description:
> > Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
> > 
> > This is a reland of a0adf3d4409036d095480e9bfa0fc06990362f84
> > 
> > Original change's description:
> > > Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
> > > 
> > > This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
> > > 
> > > Original change's description:
> > > > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> > > >
> > > > Bug: chromium:396091
> > > > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> > > > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> > > > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> > > > Cr-Commit-Position: refs/heads/master@{#29083}
> > > 
> > > Bug: chromium:396091
> > > Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
> > > Commit-Queue: Tommi <tommi@webrtc.org>
> > > Reviewed-by: Tommi <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#29655}
> > 
> > Bug: chromium:396091
> > Change-Id: I47525911095fabc6cee613d03b0d83134b95b084
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158900
> > Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30032}
> 
> Bug: chromium:396091
> Change-Id: I03702c8ea935bb5fe1797defda1ba6b279b95217
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165724
> Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
> Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#30461}

TBR=zijiehe@chromium.org,jamiewalch@chromium.org,tommi@webrtc.org,julien.isorce@chromium.org,sergeyu@chromium.org,tommi@chromium.org,trevor.axiom@gmail.com,jonringle@gmail.com,justin.franco@arterys.com

Change-Id: I1aa5092d90e4067533b639656ac822a6f920de76
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:396091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168242
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30464}
2020-02-06 08:21:42 +00:00
703a5d76d9 Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""
This is a reland of af51be7869994a299451e22e6382ae641767b26d

Original change's description:
> Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
> 
> This is a reland of a0adf3d4409036d095480e9bfa0fc06990362f84
> 
> Original change's description:
> > Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
> > 
> > This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
> > 
> > Original change's description:
> > > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> > >
> > > Bug: chromium:396091
> > > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> > > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> > > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> > > Cr-Commit-Position: refs/heads/master@{#29083}
> > 
> > Bug: chromium:396091
> > Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29655}
> 
> Bug: chromium:396091
> Change-Id: I47525911095fabc6cee613d03b0d83134b95b084
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158900
> Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30032}

Bug: chromium:396091
Change-Id: I03702c8ea935bb5fe1797defda1ba6b279b95217
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165724
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30461}
2020-02-05 20:03:19 +00:00
72859e5e15 Make RtpEncodingParameters to not reverse active flags order
Bug: webrtc:11319
Change-Id: If63db02d282ee622c12405f85c0fbae1ba13fcb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168196
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30459}
2020-02-05 17:36:26 +00:00
02b17a5507 Add helper to calculate frame dependencies based on encoder buffer usage
Bug: webrtc:10342
Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 16:19:10 +00:00
2fca97168b Delete header file mock_vcm_callbacks.h
Move definitions of mock classes to the only user, the unit tests for
the deprecated class vcm::VideoReceiver.

Bug: webrtc:7408
Change-Id: I05e38ed8ebbe615bb2db0b631ec914773fb0a520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168182
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30451}
2020-02-04 14:20:46 +00:00
f5d877847f Reland "[VP9] Shift spatial layers on RTP level to always start from 0."
This reverts commit 2181228624d1be60903c4e3352629290b9c3b27a.

Reason for revert: Reland without changes as it's not the root cause.

Original change's description:
> Revert "[VP9] Shift spatial layers on RTP level to always start from 0."
> 
> This reverts commit 2e73a3d1e9298da6a010cd638f08f36abeba11e2.
> 
> Reason for revert: Fuzzer found some issues.
> 
> Original change's description:
> > [VP9] Shift spatial layers on RTP level to always start from 0.
> > 
> > This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
> > about enabled layers from encoder to packetizer.
> > 
> > Bug: webrtc:11319
> > Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30428}
> 
> TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:11319
> Change-Id: I27a7e82737fa604b8ab769ce6503fa93e46f4e86
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168123
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30448}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org

Change-Id: Ibcd9b6a075ee08c9402de8b0b9d99d77bf59d0ef
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11319
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168185
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30450}
2020-02-04 10:06:44 +00:00
2181228624 Revert "[VP9] Shift spatial layers on RTP level to always start from 0."
This reverts commit 2e73a3d1e9298da6a010cd638f08f36abeba11e2.

Reason for revert: Fuzzer found some issues.

Original change's description:
> [VP9] Shift spatial layers on RTP level to always start from 0.
> 
> This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
> about enabled layers from encoder to packetizer.
> 
> Bug: webrtc:11319
> Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30428}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11319
Change-Id: I27a7e82737fa604b8ab769ce6503fa93e46f4e86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168123
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30448}
2020-02-03 14:15:44 +00:00
a118702566 in RtpFrameReferenceFinder VP9 case validate number of references in gof
number of references can't be invalid if gof was correctly parsed
from a vp9 packet, but RtpFrameReferenceFinder still better be
protected from the invalid data.

Bug: chromium:1048013
Change-Id: I548f5c87199421b7736409cbcacbec760ad799ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168124
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30444}
2020-02-03 10:31:38 +00:00
545c53e22f In RtpFrameReferenceFinder VP8 clean not yet received before filling it
To make it generally faster, specially in case of very large picture id gaps.

Bug: None
Change-Id: Ib0c49c17bd1281190da986def43bea8fc3440c0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168055
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30438}
2020-01-31 18:10:48 +00:00
8ad9e74d62 Removing deprecated legacy noise suppressor
This CL removes the code for the deprecated legacy noise.

Bug: webrtc:5298
Change-Id: If287d8967a3079ef96bff4790afa31f37d178823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167922
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30434}
2020-01-31 07:14:25 +00:00
be99ee8f17 Add more options for tuning the RobustThroughputEstimator through field trial.
Bug: webrtc:10274
Change-Id: I94a8c200947c66277d67812bc1d0acc9e1f40e7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168045
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30432}
2020-01-30 19:05:56 +00:00
2e73a3d1e9 [VP9] Shift spatial layers on RTP level to always start from 0.
This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
about enabled layers from encoder to packetizer.

Bug: webrtc:11319
Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30428}
2020-01-30 16:07:32 +00:00
670af2692e in RtpSenderVideo add support for writing DependencyDescriptor header extension
Bug: webrtc:10342
Change-Id: I12cca9c5e1606338bb914e58e13d268bbc6961f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166532
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30427}
2020-01-30 16:06:27 +00:00
95cb56bd89 Add extra input validation to RtpFrameReferenceFinder for codec-specific cases
wrap ids before unwrapping: should be noop for ids arrived from the
network, but avoids DCHECKs for ids arrived from fuzzer.

for vp9 double check number of references doesn't exceed maximum.
for vp8 drop key frames for non-zero temporal id.
for general by seqnum code path do not set last_picture_id_:
it is not used there, but may confuse vp8 codepath.

as a slight speed up avoid copying RTPVideoTypeHeader for vp8 and vp9.

Bug: chromium:1046995, chromium:1047024, chromium:1047095, chromium:1047165, chromium:1047190
Change-Id: I1ab0833d32e2c023cbf5e3cfcc9e74f1c558e44b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168040
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30426}
2020-01-30 15:04:05 +00:00
c31a4ec66a Disable opus tests to allow upgrade to opus 1.3
The upgrade to opus 1.3 is easier to carry out while the opus
bitexactness tests are temporarily disabled.

Bug: webrtc:11325
Change-Id: I96eecdbc93a01da88b92ae7f6473034c9795f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167726
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30425}
2020-01-30 14:57:15 +00:00
bef818d4d9 Default disables legacy overhead calculation.
This ensures that overhead calculation is correct by default when
enabling the WebRTC-SendSideBwe-WithOverhead field trial.

We keep the legacy mode to allow downstream projects already relying on
WebRTC-SendSideBwe-WithOverhead to preserve the current behavior.

Bug: webrtc:6762
Change-Id: I84369c760d59345a48ec352997dbed6d2db21d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167862
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30424}
2020-01-30 14:06:07 +00:00
5dca3f1336 Add floating point support for writing and reading wav files
This CL adds support for reading and writing floating point
wav files in WebRTC. It also updates the former wav handling
code as well as adds some simplifications.

Beyond this, the CL also adds support in the APM data_dumper
and in the audioproc_f tool for using the floating point wav
format.

Bug: webrtc:11307
Change-Id: I2ea33fd12f590b6031ac85f75708f6cc88a266b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162902
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30423}
2020-01-30 13:38:19 +00:00
190539717b Remove unused NextFrame function from FrameBuffer.
Also updated FrameBuffer unittests to use the GlobalSimulatedTimeController.

Bug: webrtc:7408, webrtc:9378
Change-Id: I8ade27492f66cdd8950b38f5f4a268714dbc35fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164536
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30422}
2020-01-30 12:54:08 +00:00
e7bc3a3477 Reland "Adds trial to use correct overhead calculation in pacer."
This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b.

Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883).

Original change's description:
> Revert "Adds trial to use correct overhead calculation in pacer."
> 
> This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7.
> 
> Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict.
> 
> Original change's description:
> > Adds trial to use correct overhead calculation in pacer.
> > 
> > Bug: webrtc:9883
> > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30399}
> 
> TBR=sprang@webrtc.org,srte@webrtc.org
> 
> Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9883
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30409}

TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30414}
2020-01-29 18:45:16 +00:00
c3eb9fd49f Reland "Reland "Only include overhead if using send side bandwidth estimation.""
This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
2020-01-29 18:42:34 +00:00
4356490b7b Revert "Reland "Only include overhead if using send side bandwidth estimation.""
This reverts commit 086055d0fd9b9b9efe8bcf85884324a019e9bd33.

Reason for revert: Causes some perf regressions.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
> 
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
> 
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> > 
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
> 
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11298
Change-Id: Id38de92ac25a1ce9a1360f0e37f65747d4cfb31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30411}
2020-01-29 16:38:57 +00:00
740ed473dc Add 444 support for vp9 decoder wrapper.
Chromting is trying vp9 444 to have better color. This fix is needed to decode 444 properly.

Bug: webrtc:11326
Change-Id: I4498930591d8876af9f6b7238a8c9fe450ecbfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166220
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30410}
2020-01-29 16:32:10 +00:00
7affd9bcbb Revert "Adds trial to use correct overhead calculation in pacer."
This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7.

Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict.

Original change's description:
> Adds trial to use correct overhead calculation in pacer.
> 
> Bug: webrtc:9883
> Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30399}

TBR=sprang@webrtc.org,srte@webrtc.org

Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30409}
2020-01-29 15:29:04 +00:00
26b4cb3fc5 Detach RtpFrameReferenceFinder from RtpGenericFrameDescriptor
To allow to use the RtpFrameReferenceFinder with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: Ib60a505a714993862a008300aa64d0bb835c3377
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167361
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30407}
2020-01-29 12:36:10 +00:00
73a5e916a9 Remove task_queue dependency for QualityScaler
This allows for the possiblity to move the QualityScaler
out of the VideoStreamEncoder in the future.


Bug: webrtc:11222
Change-Id: I1d563cf08791e27ff5065ce90bcb150a7974d868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167534
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30406}
2020-01-29 12:14:10 +00:00
182c2b8334 Expose run function to NetEqSimulator
Bug: webrtc:11005
Change-Id: I84f01536b40ba17e66877cdced194e05b882b5c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167537
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30405}
2020-01-29 11:55:05 +00:00
97ffbefdab Pass and store PacketBuffer::Packet by unique_ptr
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.

Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
2020-01-29 11:48:55 +00:00
48be482d73 Fix spelling.
Bug: None
Change-Id: Id281fe3d58bd5a8651b299b426353524085dd876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167536
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rikard Lundmark <lundmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30402}
2020-01-29 10:50:14 +00:00
71a77c4b3b Adds trial to use correct overhead calculation in pacer.
Bug: webrtc:9883
Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30399}
2020-01-29 09:39:40 +00:00
b6bf0b2546 Pass picture_id from generic packetizer through codec-specific field
To free up RtpVideoHeader::generic field for codec agnostic details
from an rtp header extension.

Bug: webrtc:10342
Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30396}
2020-01-28 19:26:28 +00:00
260c788d77 AEC3: Added multi-channel support for the capture delay functionality
This CL adds the missing support for multi-channel in the code that
provides an optional and configurable delay to be added to the
microphone signal.

The CL also makes the creation of the delay object conditional on the
need for that support (this is important since this adds a significant
heap memory footprint)

Bug: webrtc:11314,chromium:1045910
Change-Id: I92d577e31af830945fe9d5ca2032000aad4266be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167525
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30392}
2020-01-28 15:39:26 +00:00
086055d0fd Reland "Only include overhead if using send side bandwidth estimation."
This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

Bug: webrtc:11298
Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30390}
2020-01-28 10:36:39 +00:00
c709412c76 Revert "Only include overhead if using send side bandwidth estimation."
This reverts commit 8c79c6e1af354c526497082c79ccbe12af03a33e.

Reason for revert: Introduced a Bug that can happen if the include overhead state changes between pushing and poping a packet from the pacer packet queue.

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

Change-Id: I0cacbc26408b7bec5bc3855a628e62407c081117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30383}
2020-01-27 15:09:49 +00:00
8c79c6e1af Only include overhead if using send side bandwidth estimation.
Bug: webrtc:11298
Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30382}
2020-01-27 14:19:54 +00:00
ff0e4dbd1f Reland "Send absolute capture time through audio coding module."
This is a reland of 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a

Original change's description:
> Send absolute capture time through audio coding module.
>
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

Bug: webrtc:10739
Change-Id: I10086d239ad3f1efb8485098bf3b0ad23110962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167213
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30380}
2020-01-27 13:18:27 +00:00
6c9bc396e9 Cleanup log formatting in modules/audio_processing
Bug: None
Change-Id: I47177530d8a85d7b2f143081de71f5a3bf8ec354
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166041
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30379}
2020-01-27 09:42:56 +00:00
f3886aea86 Include cursor rects in updated_region.
DesktopAndCursorComposer adds the cursor image to the desktop, but does
not change the updated_region, so it generally doesn't encode correctly
unless the mouse is moving over a region that is changing. This CL
extends the updated region to include the union of the old and new
cursor rects, with an optimization for the case where the cursor has
neither moved nor changed.

Bug: chromium:1043325
Change-Id: I52076c96528820833fda6aa95f5b1fbc0f613909
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166545
Reviewed-by: Sergey Ulanov <sergeyu@google.com>
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30374}
2020-01-24 20:16:58 +00:00
b039c30157 Reland "Change log level of AEC3 buffer info to VERBOSE"
This is a reland of 48148dc840f66c5f6adc5e2ba01c15104e0a9bab

Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
>
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
>
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}

Bug: webrtc:11278, webrtc:11295
Change-Id: I8e6f11457e283c83cae5581adcacdc4d3b5431bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167360
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30372}
2020-01-24 12:58:08 +00:00
b18c4eb0a9 Add parameterization for three multi channel AEC3 unit tests
Bug: webrtc:11295
Change-Id: I478aa02908c494cf9609db00021438a59a132b66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167202
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30370}
2020-01-24 12:26:46 +00:00
159c414ff8 Detach LossNotificationController from RtpGenericFrameDescriptor
To allow to use the LossNotificationController with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: I5ac44dc5549dfcfc73bf81ad1e8eab8bd5dd136e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166166
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30369}
2020-01-24 11:53:28 +00:00
88636c6dac Improvements for NetEqControllers
When creating a NetEqController it can be useful to have access to a
webrtc::Clock*. Also, NetEqControllers should have access to the
contents of the sync buffer when making decisions.

Bug: webrtc:11005
Change-Id: I7fdba75ce661b2ace52458620a8c1f3c990e5ac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167208
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30368}
2020-01-24 11:39:52 +00:00
760fd52494 Replace MockAudioDeviceModule mock refcounting with real refcounting
Bug: webrtc:11308
Change-Id: Ic55ec2c4b45f8fc709fe1348556bdeea6202e7a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166580
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30366}
2020-01-23 19:04:58 +00:00
4175914f41 Revert "Send absolute capture time through audio coding module."
This reverts commit 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a.

Reason for revert: failing upstream tests

Original change's description:
> Send absolute capture time through audio coding module.
> 
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

TBR=danilchap@webrtc.org,ossu@webrtc.org,minyue@webrtc.org,chxg@google.com

Change-Id: Ia36b9ae899563c9afd8612ffd83871b8a5778a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167212
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30364}
2020-01-23 16:21:06 +00:00
48655cfdbf Send absolute capture time through audio coding module.
Bug: webrtc:10739
Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30363}
2020-01-23 16:06:12 +00:00
3c7e4dd85f Revert "Change log level of AEC3 buffer info to VERBOSE"
This reverts commit 48148dc840f66c5f6adc5e2ba01c15104e0a9bab.

Reason for revert: Causing tests to timeout, see bugs.webrtc.org/11295

Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
> 
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
> 
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}

TBR=saza@webrtc.org,peah@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11278
Change-Id: I283648a6d4d58cfe7af7a646d915122207883007
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30357}
2020-01-23 10:28:25 +00:00
5bb9adcb08 Add absolute capture time to video sender path.
Bug: webrtc:10739
Change-Id: I2bbef7275ae065312ad86daaecc773c0ab36a684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167061
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30344}
2020-01-22 13:09:28 +00:00
39c8350613 Reduce the complexity of the multichannel echo subtractor test
This CL reduces the complexity of the Subtractor.ConvergenceMultiChannel
test by
1. Slightly reducing the amount of tested combinations for the non-debug
   mode.
2. Drastically reduce the amount of tested combinations for the debug
   mode.


Bug: webrtc:11295
Change-Id: I56bfa4a1463d26e5217b6a4d7f2ef54de7aab512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166529
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30343}
2020-01-22 11:39:07 +00:00
d74c56fcd0 Add absolute capture time to audio sender path.
WebRTC prototype:
https://webrtc-review.googlesource.com/c/src/+/158520

Bug: webrtc:10739
Change-Id: I07b7a60602b41dc04292a91923e878a8d753486f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161732
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30335}
2020-01-21 13:06:18 +00:00
ccbe95fd8a Reformat GN files.
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.

Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.

CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn

Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:13:11 +00:00